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| 00:12.59 | robbie` | <PROTECTED> |
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| 00:46.40 | Khratos | good evening |
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| 01:07.39 | Kobaz | yeap yeap |
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| 02:17.38 | GreatSUN | rehi all |
| 02:18.15 | GreatSUN | someone alive? |
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| 02:42.49 | Thedr | Hi guys, Does anyone know where IP phones pull the caller ID for the party they are calling from? Ie if a user dials a number 123456789 it will come up with Calling John Doe, and I'm sure its not on the phones local address book |
| 02:43.59 | WiretapWork | Thedr, depends on the phone, but in most cases, addressbook |
| 02:45.00 | Thedr | The phone in question doesn't have a phonebook |
| 02:46.01 | WiretapWork | interesting |
| 02:46.09 | WiretapWork | Asterisk doesn't supply the details as far as I know |
| 02:46.57 | Thedr | to be a tad more specific, Phone 1 wants to call Phone 2, Phone 1 enters phone 2s number, 123456, once phone 1 hits dial the number, the display on phone 1 changes from calling 123456 to calling Phone 2. Phone 1 does not have an address book |
| 02:47.13 | Thedr | its very odd |
| 02:47.38 | Thedr | I have updated the name for phone 2 to phone 3 but it is still displaying as phone 2 |
| 02:47.48 | WiretapWork | I have no idea |
| 02:47.58 | Thedr | I thought the phone might be pulling the info from asterisk somewhere |
| 02:51.04 | Thedr | np, thanks for giving it some thought though |
| 02:58.11 | kaldemar | Thedr: it's a feature called COLP, the phone gets it from asterisk. |
| 02:59.17 | Thedr | excellent, do you know where it pulls the information from? as its showing the incorrect information |
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| 03:00.13 | kaldemar | Thedr: the corresponding channel configuration file or the callee. |
| 03:01.05 | kaldemar | or dialplan of course if set there. |
| 03:01.11 | WiretapWork | oooh |
| 03:01.26 | WiretapWork | I wanna set that up :P |
| 03:01.33 | WiretapWork | hopefully the pile of phones sitting next to me supports it |
| 03:02.50 | kaldemar | func CONNECTEDLINE is used in dialplan. |
| 03:03.21 | WIMPy | WiretapWork: You don't need to set up anything, usually. |
| 03:03.38 | WiretapWork | WIMPy, hmm, well my phone doesn't pull it |
| 03:03.42 | WIMPy | Unless you want to show something different from the caller ID of the called party. |
| 03:04.23 | WIMPy | I think most phones don't support it, yet. |
| 03:04.31 | WiretapWork | ah |
| 03:04.32 | WiretapWork | balls |
| 03:04.37 | WIMPy | Mind you, SIP has become quite a big mess. |
| 03:04.44 | WiretapWork | would have loved for my 7970 nad 7912s to support it |
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| 03:23.50 | GreatSUN | moin WIMPy |
| 03:24.27 | WIMPy | Moin |
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| 03:35.49 | justdave | I have two servers, both running 1.8.4.1, with an IAX trunk between them |
| 03:36.12 | justdave | I'm getting a lot of packet loss on the iax connections, but only in one direction (everything going the other way is crystal clear) |
| 03:36.30 | justdave | and the only thing reporting packet loss is the "iax2 show netstats" within asterisk |
| 03:36.45 | justdave | traceroute and ping and whatnot aren't detecting any packet loss between the two servers |
| 03:39.46 | justdave | anyone have any ideas if there's possible settings mismatches within the iax config that would cause that kind of thing? |
| 03:40.09 | justdave | comparing the [general] section and the two hosts' entries for each other, I'm not finding any differences aside from IP address and hostname |
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| 03:40.30 | WIMPy | There shouldn't be. |
| 03:40.49 | WIMPy | If it's not the connection, I can only think of timing issues. |
| 03:41.30 | WIMPy | Have you tried using another channeltype or things like MOH or Echo() on either end? |
| 03:42.43 | justdave | hmm... one has physical hardware used by the dahdi drivers (T1/PRI), the other is just using dummy... |
| 03:48.55 | justdave | I'm not sure what would affect timing if it's using dahdi_dummy |
| 03:49.03 | justdave | but that does seem like a logical place to look |
| 03:49.20 | WIMPy | Is it real hardware or any VM stuff? |
| 03:49.33 | justdave | real hardware |
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| 03:49.39 | WIMPy | good |
| 03:50.09 | WIMPy | And you're using dahdi dummy for timing? |
| 03:50.17 | justdave | on one end |
| 03:50.32 | WIMPy | ja, the oter one is obvious :-) |
| 03:50.40 | justdave | the other end is using wcte12xp |
| 03:50.59 | WIMPy | Not too likely to go wrong. |
| 03:51.46 | WIMPy | so, |
| 03:51.57 | WIMPy | Have you tried using another channeltype or things like MOH or Echo() on either end? |
| 03:52.16 | justdave | I'm playing MOH right now and watching it drop packets :) |
| 03:52.59 | WIMPy | I did have a problem that only appeared with certain channeltypes in the very beginning of 1.8. |
| 03:53.40 | justdave | It was doing this on 1.4, too |
| 03:53.42 | WIMPy | IIRC sip-iax-sip worked, but sip-iax-isdn didn't. |
| 03:54.06 | WIMPy | In that case I'd suspect the connection. |
| 03:54.09 | justdave | I'm using a SIP softphone, and IAX between the two servers |
| 03:54.24 | justdave | I'll set up a SIP link and see what happens |
| 03:54.45 | WIMPy | Use some nework monitoring tool to count coming and going packets on both sides. |
| 03:55.30 | WIMPy | Or try another codec. |
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| 03:58.02 | WIMPy | waves good night |
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| 04:15.51 | luke0512 | good morning or as we say in german moin or moinsen |
| 04:30.01 | ChannelZ | Fahrvergnugen |
| 04:41.53 | WIMPy | can't sleep :-( |
| 04:42.17 | WIMPy | So moin again. |
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| 05:16.48 | gruvfunk | waves g'nite |
| 05:17.15 | irroot | good morning grufunk |
| 05:32.26 | luke0512 | morning again |
| 05:33.04 | luke0512 | voicemail is now working...just reboot today and now it works |
| 05:37.08 | ChannelZ | It's a good day to ignore your phone and let it ring |
| 05:37.37 | luke0512 | yeah |
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| 06:00.06 | schmidts | good morning |
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| 06:06.10 | irroot | cerberus_za hehe keep it local nice to see a fellow countryman in these parts |
| 06:08.29 | kleszcz | morning |
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| 06:29.26 | BeeBuu | ~book |
| 06:29.27 | infobot | For more information about the Asterisk book, see ~thebook |
| 06:29.32 | BeeBuu | ~thebook |
| 06:29.32 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 06:34.18 | schmidts | we should name it thebible instead of thebook ;) |
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| 06:37.38 | Maliuta | there is already a bible ... it's by K&R |
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| 06:48.47 | schmidts | isnt * ANSI and not K&R? |
| 06:53.08 | ChannelZ | He's talking about the C book |
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| 06:54.41 | GreatSUN | re |
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| 07:00.01 | ChannelZ | rere |
| 07:00.33 | schmidts | ChannelZ i know thats why i said asterisk depens on ANSI not on K&R ;) |
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| 07:10.41 | zkn | Hello |
| 07:11.09 | ChannelZ | oHell |
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| 07:19.45 | zkn | ok, my head is empty from getting the configs and firewalls right.. what could be the issue that sound for a SIP call to SIP trunk travels out but is not received back? |
| 07:20.01 | ChannelZ | incoming port blocked |
| 07:20.20 | ChannelZ | or that outgoing port on the remote end blocked |
| 07:21.09 | ChannelZ | Your * requests the remote end send RTP to a port somewhere in the range listed in rtp.conf - so that range needs to be open on your end, but it's possible the remote end is not able to send to you at that port |
| 07:22.18 | GreatSUN | does someone have any idea what mistake I could have made in configuration if incoming calls work, internal sip-calls work, but external calls to other phonenumbers are established and hung up within the first second? |
| 07:23.12 | GreatSUN | my setup is alike <asterisk1> <-> VPN-Tunnel through ADSL-Line <-> <asterisk2> <-> <voip-provider> |
| 07:23.50 | kaldemar | zkn: are you talking about signaling in SIP or audio in RTP? is it a no audio problem or does the call even get set up? |
| 07:23.56 | ChannelZ | Start investigating by turning on SIP debug and see if one end is specifically terminating for some reason but who knows with the VPN, it could be any number of things |
| 07:24.43 | GreatSUN | the funny thing is |
| 07:24.52 | GreatSUN | after the connection is made |
| 07:25.25 | GreatSUN | and the response of asterisk1 has taken the call and hung up |
| 07:25.37 | zkn | in the firewall i have opened UDP: 4569, 5060-5061, 10000-20000, TCP: 5060-5061, 5038.. when I call to, say, my cellphone number and answer, then I can hear everything on my cell, but at the other end I cannot hear anything |
| 07:25.38 | luke0512 | i'm out...bye |
| 07:25.56 | GreatSUN | the phones are ringing again and the other side just hears nothing |
| 07:27.10 | kaldemar | zkn: sounds like your asterisk is behind a NAT and you don't have all the appropriate NAT settings in sip.conf. see that you have nat=yes, externaddr and localnet set under [general]. |
| 07:27.40 | ChannelZ | Is the * box the same as the firewall, or behind it with a LAN IP? |
| 07:28.52 | tuxx- | zomg |
| 07:28.56 | tuxx- | no more skype support? |
| 07:28.59 | tuxx- | fscking microsoft :P |
| 07:29.08 | zkn | * box has an external IP, it does not get IP for LAN from firewall/router DHCP, so i'm not sure if externadd and localnet will make anydifference |
| 07:29.12 | ChannelZ | well, for 2 years but after that, who knows |
| 07:29.47 | zkn | and I've tried both with nat=yes and nat=no in [general], no effect |
| 07:30.19 | ChannelZ | those really affect peers that connect to you, not you yourself |
| 07:30.39 | zkn | hmm.. |
| 07:31.06 | kaldemar | tuxx-: suprised? |
| 07:31.54 | ChannelZ | If * is behind NAT it's important to set externip and localnet accordingly, but that doesn't sound like your problem if your box really has the external IP |
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| 07:32.12 | zkn | yes, it has external IP |
| 07:32.41 | kaldemar | zkn: pastebin CLI output of a call with sip debug and verbosity enabled. |
| 07:32.46 | tuxx- | kaldemar: not really |
| 07:32.46 | zkn | ok |
| 07:32.56 | tuxx- | microsoft taking over skype, bad things are gonna happen :P |
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| 07:40.54 | schmidts | tuxx- bad things allready happend: http://now.eloqua.com/es.asp?s=491&e=162556 |
| 07:41.56 | zkn | so, here we go: http://pastebin.com/d5TtRxNX |
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| 07:42.53 | kaldemar | zkn: set directmedia=no for your local phones. |
| 07:43.02 | zkn | okay |
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| 07:46.00 | zkn | kaldemar,just out of curiosity, what did you check in that output to make this suggestion? |
| 07:46.30 | zkn | anyway, directmedia=no, sip reload, test call, no changes |
| 07:48.39 | kaldemar | zkn: the eyebeam has a private address. if asterisk sent re-invites, there would be a private address in the SDP from the eyebeam, which would break audio. |
| 07:48.58 | irroot | http://vistasucks.wordpress.com/2007/06/13/gm-vs-microsoft/ <- if M$ made cars |
| 07:49.14 | kaldemar | zkn: but that doesn't seem to be the case here. |
| 07:50.09 | cneb3000 | good morning vietnam |
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| 07:52.36 | zkn | kaldemar, i found one more directmedia in my peer template section, changed that also to "no", sip reload, still the same |
| 07:52.56 | zkn | should I provide new log? |
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| 07:59.00 | tuxx- | Hey guys, im trying to Redirect a call through AMI, this call is in the parkinglot, and when i do the redirect, i lose my audio. I get the following error: "[May 25 07:48:00] WARNING[8330]: rtp_engine.c:1209 remote_bridge_loop: Channel 'AsyncGoto/SIP/Audiocodes-0000000d<ZOMBIE>' failed to break RTP bridge |
| 07:59.16 | tuxx- | am i missing something? Do i need to send some other AMI command before i redirect this channel? |
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| 08:12.09 | Dovid | j #asterisk-il |
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| 08:19.06 | zkn | kaldemar, i think i fixed my issue |
| 08:20.17 | zkn | kaldemar, it was still something in the firewall, i disabled "Enable SIP Transformations" setting (whatever that is supposed to do anyway) and now audio was properly routed both ways for my SIP call |
| 08:22.56 | kaldemar | zkn: the good (or bad) old ALG. those are better turned off when using asterisk. |
| 08:24.25 | zkn | kaldemar, thanks for taking time to help me, appreciated! |
| 08:25.44 | Drgb_ | hello, a couple of days ago I came here asking for some help with hylafax -> iaxmodem -> asterisk, the issue was that I was able to receive faxes (and calls in general) but I couldn't send anything to an external number (internal worked). After days of coffee and headaches I thought it could've been a IAX2 related problem, so I created a new IAX2 extension and connected a softphone (zoiper). Guess what? Same result. I can only |
| 08:25.45 | Drgb_ | call internal numbers, while external calls hang some seconds with " -- DAHDI/i2/somenumber-2e is proceeding passing it to IAX2/764-6832" and fail with " -- Channel 0/1, span 2 got hangup request, cause 3". Outgoing calls get routed to a DAHDI trunk using a B410P 4 ports BRI card. SIP phones can call without any problem, the problem only affects IAX2 devices. Just to be sure, I created a IAX2 trunk to another working Aste |
| 08:25.45 | Drgb_ | risk PBX and tried to route calls through that trunk instead of the DAHDI trunk. Well, it worked. I can call external numbers through the IAX2 trunk but not through the DAHDI trunk. What am I overlooking? Any ideas? |
| 08:26.19 | Drgb_ | I can provide both failing and successful call logs if necessary |
| 08:32.49 | irroot | Drgb can also look at T38Modem / OOH323 the faxgateway code is available to get faxes out on ISDN |
| 08:34.19 | Drgb_ | irroot, thank you for the advice, but I realized that I'm facing something not specifically related to faxes. Any IAX2 outgoing call passing through the DAHDI trunk fails |
| 08:35.04 | irroot | with a BP410 and been awake now im assuming its set up for ALAW |
| 08:35.50 | Drgb_ | I tried alaw, ulaw, gsm and g726 |
| 08:36.00 | irroot | and IAX has ALAW |
| 08:36.13 | irroot | if IAX<->IAX works |
| 08:36.51 | Drgb_ | it does |
| 08:37.13 | Drgb_ | and yes, I'm currently trying with alaw, as I read it was the right choice |
| 08:37.41 | Drgb_ | (sorry but my knowledge concerning this field is not very deep) |
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| 08:56.50 | verywiseman | when asterisk start it is not record dial logs in log file ,but when i run asterisk -rvvv , it start to record , why? |
| 08:58.06 | kaldemar | dial logs? which file? |
| 08:58.54 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 08:59.11 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 08:59.12 | *** mode/#asterisk [+o file] by ChanServ |
| 09:00.15 | *** join/#asterisk dinesh___ (~dinesh@195.70.0.78) |
| 09:00.26 | *** join/#asterisk boazb (~boaz@bzq-82-80-219-90.red.bezeqint.net) |
| 09:00.43 | dinesh___ | hey folks, i am getting 3-4 notices per second about the IP 50.57.74.130 trying to connect with a wrong password on my asterisk |
| 09:01.01 | dinesh___ | is it possible to ignore it for a little while ? |
| 09:01.14 | kaldemar | block it with iptables. |
| 09:01.20 | atan | fail2ban might be of use to you |
| 09:01.21 | cneb3000 | verywiseman: look at this ---> http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf |
| 09:01.29 | Chainsaw | atan: I second that, that seems to be only way. |
| 09:01.49 | Chainsaw | dinesh___: Asterisk accepts candy from strangers and happily steps into the back of windowless vans. |
| 09:01.59 | Chainsaw | dinesh___: I recommend installing some scripts to watch over it. |
| 09:02.15 | dinesh___ | i just wonder how they got the right username |
| 09:02.36 | cneb3000 | dinesh___:guessing? |
| 09:02.42 | Chainsaw | dinesh___: Possibly because you run an outdated version or because you configured SIP wrong. |
| 09:02.50 | atan | dinesh___, what's the uername? |
| 09:02.56 | Chainsaw | dinesh___: In which case "valid account, wrong password" and "no such account" send different responses. |
| 09:03.29 | dinesh___ | well actually maybe my server is trying to connect to that ip |
| 09:03.30 | atan | ^ that's awful with regard to security |
| 09:03.42 | Chainsaw | atan: It is, which is why it got changed. |
| 09:03.55 | kaldemar | dinesh___: see that you don't have alwaysauthreject=no in sip.conf. |
| 09:04.11 | Chainsaw | kaldemar: Thanks, that's the one I meant. |
| 09:04.35 | dinesh___ | handle_request_register: Registration from '"home" <sip:home@my_server_ip>' failed for '50.57.74.130' - Wrong password |
| 09:05.01 | dinesh___ | and home is the username of my asterisk sip server |
| 09:05.53 | kaldemar | actually, the default value for alwaysauthreject was "no" before 1.8. if you're using a pre 1.8 version, put alwaysauthreject=yes in sip.conf. |
| 09:06.01 | atan | iptables -A INPUT -s 50.57.74.130 -j DROP |
| 09:06.32 | kaldemar | dinesh___: usernames like "home" are not too hard to guess... |
| 09:06.46 | cneb3000 | dinesh____: Here's an interesting article about asterisk security I read a while back.. should still be mostly up to date ---> http://blogs.digium.com/2009/03/28/sip-security/ |
| 09:07.02 | kaldemar | http://svn.digium.com/svn/asterisk/tags/1.8.4.1/README-SERIOUSLY.bestpractices.txt |
| 09:07.10 | cneb3000 | ^^^ also ---> http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood/ |
| 09:07.17 | cneb3000 | lots of night time reading for everyone :) |
| 09:07.29 | dinesh___ | oh |
| 09:07.39 | dinesh___ | because asterisk responds a difference erorr message to the client |
| 09:07.47 | dinesh___ | if it fails because of a wrong username or a wrong password |
| 09:07.58 | dinesh___ | so that's why it's easy to guess "home" |
| 09:08.08 | Chainsaw | dinesh___: I could have sworn I told you that 5 minutes ago, yes. |
| 09:08.52 | kaldemar | dinesh___: also because "home" is a simple name that is likely to exist in every dictionary that is used for attacks. |
| 09:08.57 | dinesh___ | okie dokie, so i'll set that option |
| 09:09.00 | dinesh___ | and hcange the username |
| 09:09.01 | cneb3000 | other common usernames... 1234 and test |
| 09:09.11 | dinesh___ | and read the security related articles |
| 09:09.21 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 09:09.49 | atan | I'd setup a honeypot for usernames like '0' and such so when people connect I can block them off. |
| 09:10.38 | cneb3000 | atan: that's a good idea. espiecally if it automatically blocked them. |
| 09:10.55 | dinesh___ | anyone knows the goal of those robots? |
| 09:11.01 | cneb3000 | if it could actually let them register. then you phone them and play some sort of abusive audio file. that'd be good. |
| 09:11.03 | atan | That's the idea. Let them connect, even let them try to dial out. Snag the number. Add the number to a no-call list. |
| 09:11.08 | atan | If anyone calls that number, ban them. |
| 09:11.10 | dinesh___ | is it like to make free "spamming" phone calls? |
| 09:11.16 | cneb3000 | ^^^ mostly yes |
| 09:11.34 | cneb3000 | i read an article about a guy who had his box tapped over night, and ran up a bill of a couple of thousand dollars |
| 09:11.37 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 09:11.55 | cneb3000 | ..i'll try to find it. |
| 09:12.01 | atan | I had mine run up a few hundred overnight but it turns out it was legit. Client was calling Barbados. |
| 09:12.03 | dinesh___ | glad i'm only using prepaid services |
| 09:12.11 | atan | Fucking $0.30 per minute. Daywm. |
| 09:12.32 | irroot | know of a company was nailed R100k about 10000eu over a weekend |
| 09:12.54 | dinesh___ | but still , i have a hard time seeing who needs to make so many phone calls |
| 09:13.00 | boazb | 10k is still getting out lucky |
| 09:13.01 | irroot | all to somalia/DRC/Nigeria |
| 09:13.11 | Chainsaw | It's generally people from Palestine or Israel. |
| 09:13.28 | boazb | thats very untrue |
| 09:13.37 | irroot | there is a part of johannesburg that is known for there "internet cafes" |
| 09:13.42 | Chainsaw | boazb: They're the ones hammering on my box. |
| 09:13.54 | Chainsaw | boazb: And when a user set a weak password, they called a disposable Israeli cellphone. |
| 09:13.56 | cneb3000 | dinesh____:there was scam a while back in the UK. you would buy a premium rate number, tap other peoples trunks and blast calls to it |
| 09:14.17 | irroot | its mostly Congolese / Nigerian / Somalian refugees |
| 09:14.36 | dinesh___ | yep that's smart, otherwise you have to setup a "phone company" that uses illegimitate sip trunks, much harder to do |
| 09:14.37 | irroot | they "tap" systems and bridge calls to home |
| 09:14.49 | Chainsaw | boazb: After that they seemed to want to DoS a number in Jordan. |
| 09:15.07 | boazb | Chainsaw: if you dont mind i'd like to have those test numbers |
| 09:15.29 | Chainsaw | boazb: And why is that? |
| 09:15.43 | boazb | Analysis, blacklisting, |
| 09:16.38 | Chainsaw | boazb: You're a random, anonymous person on the internet. I'm not averse to sharing such data, but I would have to know who you are. |
| 09:17.01 | cneb3000 | Chainsaw: Can I have it? I promise not to hax you. |
| 09:17.04 | cneb3000 | ;) |
| 09:18.04 | Chainsaw | cneb3000: Same answer. However boazb is connecting through Bezeq, so a legitimate local interest may exist. Not so sure about you. |
| 09:18.05 | boazb | Well, i think it would generally be in the interest of everyone to have these numbers and to black list them so I dont see any harm in sharing this data since its not really sensative to you |
| 09:18.12 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 09:18.16 | boazb | either way you can look me up: boaz bechar |
| 09:18.19 | dinesh___ | i was given on multiple occasion root passwords of boxes on irc, in order to help ;) |
| 09:19.20 | dinesh___ | but that's off topic, just saying that some people take huge risks |
| 09:20.13 | boazb | from what i've found the most costly attacks involve calls to destinations that are several $ per minute, ie sattelite phones, premium numbers in remote destinations... the Israel/palestine based attacks usually much less |
| 09:20.21 | Chainsaw | boazb: Very well. Disposable cell: +972 547369867 |
| 09:20.51 | Chainsaw | boazb: Followed by North Korea, +85026251229 |
| 09:20.55 | boazb | Thanks also if anyone else listening would like to share more fraud data please do: boaz@humbuglabs.org |
| 09:21.12 | Chainsaw | boazb: Followed by Bulgaria: +359999302644 |
| 09:21.29 | Chainsaw | boazb: Followed by San Marino: +37877310834 |
| 09:21.40 | boazb | thanks i am noting these |
| 09:22.12 | irroot | is going to set up CA with openssl and only accept TLS calls on the net |
| 09:22.47 | dinesh___ | i'd like to tape one of those conversations ;) |
| 09:22.53 | Chainsaw | boazb: Cook Islands was next: +68259190. Then Zimbabwe: +263912795955. Then Tonga Islands: +67658877. |
| 09:23.02 | *** join/#asterisk wonderworld (~ww@port-92-201-250-190.dynamic.qsc.de) |
| 09:23.05 | irroot | got some recordings of the fraud |
| 09:23.12 | Chainsaw | boazb: And then what they were really after, a number in Liberia (I said Libya before, sorry). They kept hammering on this one: +2314429173 |
| 09:23.45 | WiretapMac | Chainsaw: wow |
| 09:24.02 | irroot | any +27 numbers might be mine :P |
| 09:24.04 | Chainsaw | boazb: And also hammering on these two: +2314392044, +2314392045 (Liberian numbers) |
| 09:24.46 | Chainsaw | irroot: I'm on good terms with +27, just ask jkroon. |
| 09:24.53 | boazb | hmmm seems we had 972547369867 blacklisted since Sept 2010 |
| 09:25.10 | dinesh___ | wow i set alwaysauthjrect=true, and 5 minutes later the attack stopped |
| 09:25.32 | Chainsaw | boazb: This took place in Sept 2010, yes. |
| 09:25.49 | WiretapMac | dinesh___: you mean alwaysauthreject? |
| 09:27.48 | Chainsaw | boazb: The use of the account was from paltel (so, Palestine), the bruteforce was from a german IP. |
| 09:27.56 | Chainsaw | boazb: That's as detailed as I'm willing to go. |
| 09:28.17 | boazb | many thanks Chainsaw |
| 09:28.36 | *** join/#asterisk Cadey (~x@62.84.178.106) |
| 09:29.33 | dinesh___ | yes WiretapMac |
| 09:43.57 | Chainsaw | leifmadsen: 18898 appears to be ready for testing; the second iteration of the patch on there works for me. |
| 09:45.04 | Chainsaw | leifmadsen: "svn-320715-bad-event.diff" by gareth. |
| 09:47.20 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 09:47.53 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 09:53.45 | irroot | ok those of you that dont know regex well go write out 1000 times "regex is my friend" |
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| 10:17.54 | schmidts | interesting question, how much costs cuba for you? we have to pay 1 ? per minute thats around 1,4 $ |
| 10:18.13 | schmidts | only satelitte numbers are more expensive than cuba for us |
| 10:22.37 | zkn | has anyone here had experience deploying Asterisk in South Africa ? |
| 10:26.58 | irroot | zkn have 150+ systems out there |
| 10:27.15 | irroot | welcome to get intouch |
| 10:27.47 | zkn | cool |
| 10:30.09 | jacc0 | @boazbL I have some fraud info |
| 10:34.00 | jacc0 | fraud info : http://pastebin.com/Ec7WkshG |
| 10:38.49 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:52.26 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 10:53.40 | Drgb_ | Sorry for asking again, but I guess that after 2 hours my question is not very "visible" anymore. I'm having issues calling external numbers from a IAX2 internal softphone. Internal calls work fine. External calls only work from SIP devices (even though they're going through the same DAHDI trunk). Here's a full log with a working call from SIP to external, a working call from IAX2 to SIP internal, and a failing call from IAX2 t |
| 10:53.40 | Drgb_ | o external. I hope it helps: http://pastebin.us/4913 |
| 10:53.53 | ruyo | Anyone having problems compiling mISDN git on Debian Squeeze? |
| 10:53.59 | *** part/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 10:54.38 | WIMPy | ~ask |
| 10:54.38 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:54.42 | WIMPy | ruyo: ^^ |
| 10:55.32 | ruyo | True, wasn't very clear very clear, was I... |
| 10:56.50 | ruyo | I'm getting this error compiling mISDN git: http://pastebin.com/CZgK0nPu |
| 10:57.41 | ruyo | I already compiled it some time ago, so I'm guessing something changed. |
| 11:00.38 | WIMPy | I did not come across that particular one, but there's obviousely some incompatibility there. |
| 11:01.11 | WIMPy | You are trying to put misdn1 on that kernel, right? |
| 11:01.27 | ruyo | I'm using a rather slim debian install, base system only. |
| 11:01.32 | ruyo | v2.. |
| 11:02.39 | *** part/#asterisk irroot (~gregory@dsl-185-122-118.dynamic.wa.co.za) |
| 11:02.57 | Drgb_ | ruyo, what kernel version? and what mISDN version? |
| 11:03.10 | ruyo | Unless the git repo is for the v1 one |
| 11:03.14 | WIMPy | In that case a standard kernel should do, if not too old. |
| 11:03.38 | ruyo | uname -a: Linux debian 2.6.32-5-amd64 #1 SMP Mon Mar 7 21:35:22 UTC 2011 x86_64 GNU/Linux |
| 11:04.16 | *** join/#asterisk irroot (~gregory@dsl-185-122-118.dynamic.wa.co.za) |
| 11:04.41 | ruyo | WIMPy, I can't modprobe mISDN_core with the standard kernel. Means I don't have the mISDN module, right? |
| 11:04.52 | WIMPy | That should be good as it is. |
| 11:05.47 | WIMPy | Hmm, so Debian chose to leave them out? |
| 11:06.05 | ruyo | Ok, let me revert the snapshot to the post-install part just to be sure. |
| 11:08.29 | ruyo | It does load mISDN_core. :D |
| 11:09.20 | ruyo | The attempts at compile must have screwed the modules before. |
| 11:09.23 | WIMPy | Err, what, or rather where from now? |
| 11:09.40 | WIMPy | Ah, ok, so it did come with thte kernel? |
| 11:09.55 | ruyo | Debian Squeeze standard kernel does have mISDN. |
| 11:10.25 | irroot | WIMPy that is v2 and requires LCR not mISDN |
| 11:10.45 | WIMPy | Ok, so unless you have hardware that wasn't supported then, it will be fine. |
| 11:10.56 | ruyo | I still need mISDNuser, no? |
| 11:11.02 | ruyo | for the misdn_info? |
| 11:11.19 | WIMPy | Ye, he said v2 |
| 11:11.20 | *** join/#asterisk X-Rob (~Rob@eth2083.qld.adsl.internode.on.net) |
| 11:11.38 | ruyo | irroot, yes, chan_lcr instead of chan_misdn to connect to asterisk. |
| 11:11.38 | WIMPy | Yes, you need mISDN_user. |
| 11:11.53 | ruyo | Very well, I'm more motivated now. :) |
| 11:12.04 | ruyo | Thanks WIMPy. |
| 11:24.41 | tuxx- | Is it possible to send parameters to an application via an ami originate call? |
| 11:25.35 | tuxx- | http://pastie.org/1970752 i tried to do it like this, but that doesnt seem to work |
| 11:32.42 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 11:35.21 | ruyo | tuxx-, if Application is set, the Data field should be that applications's arguments. |
| 11:36.13 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 11:36.54 | tuxx- | right, so the example in pastie.org should do it? |
| 11:37.44 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 11:41.42 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 11:45.41 | ruyo | From the description it should. Maybe try using | instead of , to split the arguments. |
| 11:57.17 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:01.46 | *** part/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
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| 12:25.26 | dinesh___ | can i send incoming calls to different contexts? |
| 12:25.48 | dinesh___ | in my extensions.conf I made a section [redirect] and another one [incoming] |
| 12:25.59 | kaldemar | dinesh___: you can do what ever you want to. |
| 12:26.08 | dinesh___ | and for 2 sip numbers that I have, i would like to start at [incoming], and for the other one at [redirect] |
| 12:26.18 | dinesh___ | but i didn't find how to do it in the register => ... syntax |
| 12:26.43 | kaldemar | are they separate accounts? |
| 12:26.49 | dinesh___ | yes |
| 12:27.20 | kaldemar | with two defined peers in sip.conf, just configure a different context. |
| 12:27.20 | dinesh___ | well i did not even create an "account" for the last number, i just added a register => ... line in my sip.conf |
| 12:28.33 | dinesh___ | well that's the thing, i don't get how asterisk maps the peers in sip.conf with the register => instructions |
| 12:28.45 | kaldemar | do the calls to both numbers come from different locations? |
| 12:28.58 | dinesh___ | yes |
| 12:29.05 | dinesh___ | it's 3 different providers for 3 numbers |
| 12:29.50 | dinesh___ | hmm okaz |
| 12:30.04 | dinesh___ | maybe the problem comes from the fact that i put my register => at the very top of my sip.conf |
| 12:30.12 | dinesh___ | and then i should add them within the right context |
| 12:30.14 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:30.14 | dinesh___ | instead |
| 12:30.14 | kaldemar | then make 3 peers that have different contexts defined. |
| 12:30.37 | dinesh___ | okie i get it now i think, thanks |
| 12:30.58 | kaldemar | register statements belong under [general] |
| 12:32.11 | kaldemar | register => user:secret@provider/number <-- makes them call "number" |
| 12:34.03 | dinesh___ | yes |
| 12:34.15 | dinesh___ | but then how do i tell it to use a different context ? |
| 12:35.34 | kaldemar | dinesh___: http://pastebin.com/wsVfWDbm |
| 12:36.05 | kaldemar | you of course need other options in addition to context. |
| 12:36.35 | dinesh___ | but isn't that only for outgoing calls? |
| 12:36.43 | kaldemar | no |
| 12:37.08 | dinesh___ | i'll give it a try |
| 12:38.23 | kaldemar | basically, you'll need at least type=peer, host=provider_ip_address, insecure=port,invite along with the context definition. |
| 12:41.16 | dinesh___ | it seems that everything is broken now, since i changed my extension.conf |
| 12:41.29 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-zejngqtgqsirvred) |
| 12:41.53 | dinesh___ | i'm getting "call from '' to extension 's' rejected because extension nout found in context 'default'" |
| 12:42.38 | *** join/#asterisk QuantumSchema (~QuantumSc@rrcs-24-227-92-3.se.biz.rr.com) |
| 12:42.47 | leifmadsen | dinesh___: time to start debugging with 'sip set debug on' |
| 12:43.15 | leifmadsen | look at the INVITE and track down what it is matching on (or not matching on) and determine what you have wrong in your configuration that the matching of the peer is incorrect |
| 12:52.17 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 13:00.14 | *** join/#asterisk DerkKo (~afernande@199.255.83.50) |
| 13:03.44 | DerkKo | Im trying to add information into the to: field... i tried using SIPAddHeader but it just adds a second to field... exten => _X.,3,SIPAddHeader(To: AsteriskRecorder|channel=${CHANNEL}|OrigLegcall-id=${SIP_HEADER(Call-ID)}|") |
| 13:04.42 | DerkKo | Also this brakes the URI, im trying to append this information after the normal URI IE: To: <sip:3320@192.168.1.1>;AsteriskRecorder|channel=${CHANNEL}|OrigLegcall-id=${SIP_HEADER(Call-ID)}| |
| 13:07.32 | kaldemar | DerkKo: you can only add new headers with SIPAddHeader |
| 13:08.39 | *** join/#asterisk wonderworld (~ww@port-92-201-250-190.dynamic.qsc.de) |
| 13:08.56 | DerkKo | So the question is... How can i modify the to header ? |
| 13:09.09 | DerkKo | not modigy, basically add information to the to header |
| 13:09.35 | kaldemar | either modify sources or use something other than asterisk. |
| 13:09.39 | *** join/#asterisk jaybinks (~jaybinks@203.62.187.176) |
| 13:09.51 | DerkKo | mmmmm |
| 13:09.53 | DerkKo | VXML_URL |
| 13:10.00 | DerkKo | i think this may be the answer |
| 13:10.16 | DerkKo | http://www.voip-info.org/wiki/view/Asterisk+SIP+channels |
| 13:10.37 | DerkKo | VXML_URL |
| 13:10.38 | DerkKo | Phones running the SCCP (skinny) firmware have some support for pushing XML pages. If you want to test it, set the variable VXML_URL to point to a Cisco XML file on a web server. |
| 13:10.38 | DerkKo | This adds information to the SIP "To:" header, and it could be used for other purposes if there are other phones that can take extra information in this way. For example: |
| 13:10.57 | DerkKo | Not sure if its only for SCCP |
| 13:11.02 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:11.56 | *** join/#asterisk just187 (~just187@217.148.237.2) |
| 13:19.51 | jaybinks | hey can you guys help me test my new box ... you should be able to call sip://lenny@203.33.61.11:5060 |
| 13:23.01 | jaybinks | anyone ?? |
| 13:25.59 | just187 | ah no :-) |
| 13:27.22 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:27.48 | jaybinks | why ? |
| 13:29.16 | just187 | because my telefon doesnt have a / button |
| 13:30.24 | QuantumSchema | Good mornin' all! |
| 13:30.30 | *** join/#asterisk m_tadeu (~quassel@89.180.67.125) |
| 13:30.44 | QuantumSchema | leifmadsen: Sorry I didn't get to finish our conversation yesterday. |
| 13:30.51 | jaybinks | just187 - you can throw it in your asterisk ... :P |
| 13:30.56 | QuantumSchema | leifmadsen: I got pulled away from desk and never got to get back. |
| 13:31.51 | QuantumSchema | leifmadsen: So what I was thinking with hints was to to create a hint that tied to an agent's channel to let the queue app know if the agent was on an outbound call. |
| 13:32.40 | QuantumSchema | leifmadsen: I checked with DEVICE_STATE and tried setting the state to INUSE when an outbound call was sent but it didn't quite work. |
| 13:32.49 | m_tadeu | hi...what is the difference in asterisk behaviour between these 2 agi commands: "SET CALLERID some_id", "SET VARIABLE CDR(callerid) some_id"? |
| 13:33.08 | QuantumSchema | leifmadsen: I guess my other confusion would be how does the queue app know that the agent is in use on an outbound call. |
| 13:33.21 | QuantumSchema | leifmadsen: Maybe there is a way to go about it with out hints and DEVICE_STATE? |
| 13:34.15 | kaldemar | m_tadeu: the first changes caller id, the second changes a CDR field value. two different things. |
| 13:35.50 | m_tadeu | kaldemar: I see...so if I want to change the caller id I should both...I was figuring that setting the caller id would affect also the cdr |
| 13:36.33 | QuantumSchema | leifmadsen: is there a way to manipulate the agent status that's listed when doing a "queue show" ? |
| 13:37.35 | kaldemar | m_tadeu: callerid is not a field that exists by default. clid on the other hand is. if you want to change caller id, just change the caller id and leave CDR fields alone. |
| 13:38.25 | kaldemar | m_tadeu: and CDR(clid) is read-only anyway. |
| 13:38.49 | m_tadeu | kaldemar: I see....thanx a lot |
| 13:43.04 | Dovid | anyone know of any phones that do a re-invite after x amount of time ? |
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| 14:09.59 | skrusty | afternoon all |
| 14:10.08 | cneb3000 | heidi ho |
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| 14:10.42 | Lantizia | tzafrir_laptop, Lo are you about and got a sec? |
| 14:11.10 | tzafrir_laptop | yup |
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| 14:12.01 | Lantizia | tzafrir_laptop, cool :) i use the svn 1.8.x trunk you maintain for debian - noticed 1.8.3 now depends on dahdi but it didn't before - just wondering if that was intentional? |
| 14:12.39 | Lantizia | as in the asterisk package it builds it dependant on it |
| 14:12.43 | Lantizia | *is |
| 14:12.45 | tzafrir_laptop | asterisk-dahdi depends on dahdi . asterisk does not depend on dahdi |
| 14:14.13 | Lantizia | tzafrir_laptop, hmm lemme double check - what I've found is when I apt-get install asterisk from my repo (holding the packages your debian src package creates) - it tried to pull in dahdi |
| 14:14.56 | tzafrir_laptop | Lantizia, could you please try: --no-install-recommends ? |
| 14:15.12 | Lantizia | tzafrir_laptop, I never use recommends on servers - first thing I turn off |
| 14:17.07 | Lantizia | tzafrir_laptop, sorry I mean 1.8.4 not 1.8.3 lol |
| 14:17.37 | Drgb_ | Sorry for asking again (again). I'm having issues calling external numbers from a IAX2 internal softphone. Internal calls work fine. External calls only work from SIP devices (even though they're going through the same DAHDI trunk). Here's a full log with a working call from SIP to external, a working call from IAX2 to SIP internal, and a failing call from IAX2 to external. I hope it helps: http://pastebin.us/4913 |
| 14:18.57 | Lantizia | tzafrir_laptop, perhaps I'm going crazy - will test again with a fresh server and let you know if I find the same issue |
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| 14:21.31 | jkroon | hi guys, i clearly misunderstand something about how asterisk uses jitter buffers, and where exactly they fit into the rtp streams. my understanding was that they sit on the rx side of a link, but somehow I have a nasty suspicion they actually sit on the tx side? |
| 14:21.52 | russellb | yes ... i have a blog post about that somewhere |
| 14:22.04 | leifmadsen | indeed |
| 14:22.23 | leifmadsen | http://www.asterisk.org/node/48317 |
| 14:22.36 | russellb | http://www.russellbryant.net/blog/2007/10/09/asterisk-jitterbuffer-support-for-applications/ |
| 14:22.37 | leifmadsen | russellb: ^^^? |
| 14:22.41 | russellb | ah, which just links to that |
| 14:22.44 | leifmadsen | heh |
| 14:22.57 | russellb | well my post is something else, but references the link you put |
| 14:23.06 | jkroon | it says ast1.4 but I assume the same holds for 1.6.X and 1.8? |
| 14:23.11 | russellb | yes |
| 14:23.13 | serafie | Hmm, I just linked to those pages yesterday. |
| 14:23.19 | russellb | it hasn't changed ... until 1.10 |
| 14:23.23 | leifmadsen | dun dun dun |
| 14:23.26 | russellb | we have a new jitterbuffer method in 1.10 |
| 14:23.27 | leifmadsen | russellb: new blog post! |
| 14:23.30 | russellb | zomg |
| 14:23.58 | russellb | maybe if i stop working on books i'll feel like blogging again |
| 14:24.04 | russellb | i haven't posted much in the last year |
| 14:26.13 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59.dsl.dyn.forthnet.gr) |
| 14:26.39 | jkroon | ok, so to get a jitter buffer between SIP and DAHDI channel you actually need to enable the JB on chan_dahdi, not on chan_sip? |
| 14:27.09 | Drgb_ | could anybody help me to understand better the steps occurring during a call generated from a IAX2 extension going to a DAHDI trunk? maybe you could address me to the right direction..I'm totally lost |
| 14:27.27 | leifmadsen | Drgb_: I don't really understand the question |
| 14:27.41 | leifmadsen | it's just dialplan ... |
| 14:27.43 | Drgb_ | ok, I'll try to explain with the few concepts I know |
| 14:27.56 | leifmadsen | Dial(DAHDI/g0/${NUMBER_TO_DIAL}) |
| 14:28.23 | Drgb_ | I'm 99% sure my problem is limited to IAX2 extension going through the DAHDI trunk |
| 14:28.30 | Drgb_ | outgoing calls simply don't work |
| 14:28.36 | leifmadsen | does it work if you call another IAX2 extension? |
| 14:28.38 | Drgb_ | thei fail reporting no route to destination |
| 14:28.40 | Drgb_ | yes, it does |
| 14:28.54 | Drgb_ | and it works even if I route the call through a IAX2 trunk to another Asterisk PBX |
| 14:29.03 | russellb | jkroon: yes |
| 14:29.06 | russellb | as bizarre as that is |
| 14:29.16 | russellb | in 1.10 we have a new method that makes it work like you would expect, heh |
| 14:29.17 | leifmadsen | does eliminating IAX2 entirely (or using a SIP phone) work? It could be a configuration issue with your DAHDI channel |
| 14:29.24 | Drgb_ | yes, it works |
| 14:29.28 | jkroon | indeed! ok, that screwed me over for about two years now. |
| 14:29.29 | Drgb_ | I posted the pastebin link before |
| 14:29.33 | jkroon | will need to test over the weekend. |
| 14:29.35 | leifmadsen | Drgb_: you'll have to paste it again |
| 14:29.38 | Drgb_ | sure |
| 14:29.40 | leifmadsen | nevermind found it |
| 14:29.42 | Drgb_ | http://pastebin.us/4913 |
| 14:29.58 | leifmadsen | ugh, you're using freepbx |
| 14:30.03 | leifmadsen | that's not really a simple dialplan... |
| 14:30.15 | Drgb_ | I know, I can simplify it if needed |
| 14:30.21 | russellb | jkroon: sorry :-( |
| 14:30.22 | Drgb_ | I can temporarily exclude FreePBX |
| 14:30.53 | kaldemar | "However, it is useful to be able to de-jitter traffic in the middle of the jitterbuffer at the endpoint is not very good." <-- a minor malfunction there? |
| 14:31.28 | russellb | yeah that sentence doesn't make sense |
| 14:31.29 | Drgb_ | I could even smash a hammer on my PBX, if it can be helpful. My mental health is more important |
| 14:32.02 | raden_work | where naikorvek ? |
| 14:32.14 | leifmadsen | Drgb_: not sure why when you place a call via IAX2 you get this: 2 > Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '0799577377' ] |
| 14:32.19 | leifmadsen | (line 102) |
| 14:32.31 | leifmadsen | that's coming from the ISDN side |
| 14:33.02 | Drgb_ | it appears also in the working SIP call |
| 14:33.13 | Drgb_ | not that I know what it exactly mean |
| 14:33.16 | leifmadsen | oh I missed it there |
| 14:33.17 | Drgb_ | *means |
| 14:33.41 | Drgb_ | anyway the step at which it hangs for about 20 seconds is this... |
| 14:33.55 | Drgb_ | <PROTECTED> |
| 14:33.59 | Drgb_ | line 123 |
| 14:34.16 | Drgb_ | then it fails reporting " -- Channel 0/1, span 2 got hangup request, cause 3" |
| 14:34.20 | leifmadsen | ya I just see a DISCONNECT instead of CONNECT |
| 14:34.58 | leifmadsen | I would suggest at the least starting with something simpler |
| 14:35.08 | leifmadsen | just build a simple dialplan that places calls and see if that changes anything |
| 14:35.20 | kaldemar | Drgb_: the IAX2 call has the "Redirecting Number" part which the SIP one does not. that is the only difference in those traces in addition to the numbers. |
| 14:35.22 | leifmadsen | I'm not sure though -- I don't really use IAX2 but I don't see anything obviously wrong |
| 14:36.17 | Drgb_ | humm.. "redirecting number", thanks kaldemar |
| 14:36.20 | Drgb_ | and thank you leifmadsen |
| 14:36.41 | raden_work | leifmadsen, you have any idea why music on hold in 1.8.x does not work with aastra phones but all other brands ? |
| 14:36.47 | Drgb_ | I'll start working on a simple dialplan without FreePBX now |
| 14:37.14 | Drgb_ | then I'll come back here and randomly rant about libiax2 being broken :D |
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| 14:42.46 | kaldemar | Drgb_: there has been a change in mISDN to not send the redirecting number ie when functioning in TE mode because of deutsche telekom's network not liking it. |
| 14:43.09 | Drgb_ | kaldemar, I'm not using mISDN, I'm using a b410p with dahdi drivers |
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| 14:43.40 | kaldemar | Drgb_: may be that the particular information element is your issue, for some reason it gets created when chan_iax is used but not with chan_sip. |
| 14:43.57 | jkroon | russellb, leifmadsen - thanks for that lightbulb moment |
| 14:44.00 | kaldemar | Drgb_: yes, i noticed that. just an observation that the IE has caused problems before. |
| 14:45.21 | jaybinks | hey can you guys help me test my new box ... you should be able to call sip://lenny@203.33.61.11:5060 |
| 14:45.52 | jaybinks | sorry that should be sip:lenny@203.33.61.11:5060 |
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| 14:56.31 | QuantumSchema | Does anyone have any thoughts as to how I could manipulate an agent's status that is listed in "queue show"? Like setting an agent to "In Use" from a dial plan? |
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| 15:02.19 | leifmadsen | raden_work: I don't own any aastra phones, so no idea |
| 15:07.00 | Drgb_ | ok guys I excluded freepbx and created a very simple (2 lines) dialplan |
| 15:07.51 | Drgb_ | just a second and I'll show you the logs |
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| 15:08.44 | jaybinks | join #voip |
| 15:13.58 | Drgb_ | http://pastebin.us/4934 |
| 15:14.38 | Drgb_ | here it is, a successful call from a sip hardware phone and an unsuccessful call from a IAX2 soft phone |
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| 15:38.30 | cneb3000 | Drgb_:What's all that stuff from line 200 - about 215? |
| 15:39.18 | Drgb_ | it's my stupid dialplan not handling the end of the call |
| 15:39.37 | Drgb_ | I solved it by adding a "Congestion" action at the end of it |
| 15:39.44 | Drgb_ | I can reissue the call if needed |
| 15:40.53 | QuantumSchema | Anyone know about changing the status of a queue agent ("In Use" or "Available") via dialplan? |
| 15:42.45 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 15:43.08 | Drgb_ | (edit: I added a rule for the "h" extension, which represents the end of the call) |
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| 15:48.41 | cneb3000 | Drgb_: what about line 255. is that related to 200-215? |
| 15:49.14 | Drgb_ | let me check if it disappeared after the correction |
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| 15:51.52 | Drgb_ | it disappeared |
| 15:51.59 | Drgb_ | it was related to the "h extension" |
| 15:52.10 | Drgb_ | http://pastebin.us/4937 |
| 15:53.09 | cneb3000 | Drgb_ :ahh I see. now what about line 135:) |
| 15:53.20 | cneb3000 | sorry.... 132 |
| 15:53.24 | Drgb_ | uh, there's an incoming call in the middle of that :°D |
| 15:53.32 | Drgb_ | let me do it again, sorry |
| 15:53.37 | cneb3000 | haha no problem |
| 15:53.57 | cneb3000 | there's no thread id's.. makes it hard to follow. |
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| 15:54.57 | Drgb_ | http://pastebin.us/4938 |
| 15:55.09 | Drgb_ | thread id? how can I add it? I don't know what you're talking about, sorry |
| 15:55.26 | cneb3000 | its OK, it's like.. a unique identifier which follows the log down |
| 15:55.29 | leifmadsen | QuantumSchema: that's what DEVICE_STATE() is for |
| 15:55.36 | leifmadsen | beyond that, it's really a function of the channel |
| 15:56.24 | cneb3000 | Drgb: Hey you know the number you're dialling (0799577377?) |
| 15:56.29 | *** join/#asterisk zkn (~zkn@c83-251-131-94.bredband.comhem.se) |
| 15:56.34 | cneb3000 | Drgb: Is that a british mobile, yes? |
| 15:56.53 | Drgb_ | no, it's an italian office phone |
| 15:57.12 | QuantumSchema | leifmadsen: So would I create a hint for say "Local/1257" (the "queue show" reflects "Local/1257@agents/n"), and then use DEVICE_STATE() against that? |
| 15:57.24 | leifmadsen | hints are not device state.... |
| 15:57.25 | *** join/#asterisk deadpigeon (~deadpigeo@office.xpressamerica.net) |
| 15:57.28 | leifmadsen | I think you misunderstand what hints are for |
| 15:57.29 | Drgb_ | (owned by me, handled by another working asterisk PBX) |
| 15:57.50 | leifmadsen | hints reflect the state of the device sure, but they are not the actual device state information |
| 15:57.50 | Drgb_ | but I get the same results with any other number |
| 15:58.59 | QuantumSchema | i might truely be mistaken then. I thought Asterisk would attempt to call extensions based off of the DEVICE_STATE(). Kind of like creating a hint is kind of like allocating the location to store the state of the device. |
| 15:59.16 | leifmadsen | right, the hint is just a convenient way of monitoring the state of a device |
| 15:59.26 | leifmadsen | the device state, at least with SIP, is all automatic |
| 15:59.43 | leifmadsen | which is why it is recommended that you use SIP end points and not Agent (or anything else) end points in queues |
| 15:59.58 | leifmadsen | because only SIP has the appropriate device state stuff to make device states in queues accurate |
| 16:00.16 | leifmadsen | <PROTECTED> |
| 16:00.39 | QuantumSchema | Oh! |
| 16:01.05 | leifmadsen | so with what you're trying to figure out you can safely ignore hints entirely -- they don't help you right now |
| 16:01.59 | *** join/#asterisk tris (~tristan@2001:1868:a00a::4) |
| 16:02.57 | cneb3000 | Drgb_ : it always comes back down to this his 'no route to destination thing'.. what does the asterisk box which hosts the 0799577377 number say? |
| 16:03.22 | Drgb_ | it says nothing, it doesn't even see the call coming |
| 16:04.08 | cneb3000 | That's wierd, because you do get messages suggesting the call is about to starting ringing |
| 16:04.20 | cneb3000 | so in between the asterisk software and the other asterisk box, what is there? |
| 16:04.57 | Drgb_ | hum, the interwebz :) |
| 16:05.09 | Drgb_ | or a switch, if we want to take the shortest route |
| 16:05.41 | Drgb_ | but I repeat, it's the same with each and every number I try |
| 16:05.53 | Drgb_ | with the only distinction between valid and invalid numbers |
| 16:05.59 | Drgb_ | valid numbers return error code 3 |
| 16:06.08 | Drgb_ | while nonexistant numbers return code "38" |
| 16:06.41 | Drgb_ | so it looks like the PBX is able to start a negotiation with my provider |
| 16:06.52 | Drgb_ | but something goes wrong in the process |
| 16:07.00 | cneb3000 | Have you spoke to your provider? |
| 16:07.12 | cneb3000 | sounds like something they'll be able to help you with? |
| 16:07.29 | infernix | is anyone aware of an open source speech recognition solution that plugs into asterisk (voicexml preferably)? found a few TTS solutions but no open recognition ones |
| 16:07.30 | cneb3000 | It may be asterisk at fault, as it were, but they'll be able to tell you what it's doing wrong for them |
| 16:07.55 | cneb3000 | infernix: the good stuff - you have to pay for it :) |
| 16:08.34 | infernix | yeah but i'm trying to build a proof of concept |
| 16:08.45 | infernix | i'll pay when the project gets the go-ahead :) |
| 16:08.51 | cneb3000 | hehe ;) |
| 16:08.52 | leifmadsen | infernix: the only STT open source application I know of is Sphinx |
| 16:09.14 | infernix | i'll probably pull some strings and see if I can get a demo from one of the commercial vendors |
| 16:10.43 | Drgb_ | cneb3000, I didn't try to contact them, but I guess they wouldn't be as kind as you, and finding someone with a bit of knowledge in a (italian) call center is like winning to the lottery |
| 16:11.29 | cneb3000 | Drgb_: Wish I could help more - but not really dealt much with q.931 |
| 16:12.27 | cneb3000 | If I was you, and it looked like my carrier was rejecting my calls - i'd tell them |
| 16:12.38 | Drgb_ | cneb3000, you and the other guys have been very helpful, thank you very much, I'll idle here and write any eventual step forward |
| 16:12.54 | cneb3000 | i'd be interested! |
| 16:13.07 | Drgb_ | I would've done it if I was totally unable to call, but SIP internals work, so they could simply tell me "it's your problem, dude" |
| 16:13.09 | *** join/#asterisk jpsharp (~jsharp@74-95-145-86-Naples.hfc.comcastbusiness.net) |
| 16:13.24 | cneb3000 | haha |
| 16:13.26 | cneb3000 | techies |
| 16:13.32 | cneb3000 | you have to be harsh with them sometimes.. ¬_¬ |
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| 16:18.21 | Drgb_ | HOLY CRAP! |
| 16:18.23 | Drgb_ | IT WORKED! |
| 16:18.38 | Drgb_ | exten => _.,1,Set(CALLERID(RDNIS)=079399892) |
| 16:18.51 | cneb3000 | what, you didn't have a callerid? |
| 16:19.01 | Drgb_ | yes I do, but it seems it was the wrong one |
| 16:19.05 | Drgb_ | not the callerID |
| 16:19.10 | Drgb_ | but the "redirecting number" |
| 16:19.14 | Drgb_ | you can see it in the call log |
| 16:19.21 | cneb3000 | no wonder! |
| 16:19.25 | cneb3000 | excellent - well done :) |
| 16:19.27 | Drgb_ | there was a weird 5 digits number I didn't recognize |
| 16:19.30 | Drgb_ | lol, just luck |
| 16:20.32 | paulc | infernix: I've had a lot of success with Lumenvox and Asterisk |
| 16:21.04 | paulc | infernix: Not open source but I think there's a $50 start up pack (single channel). I can give you a number for a demo too if you like. |
| 16:27.15 | m_tadeu | is it possible to get the channel name using only the unique id? through ami, I mean |
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| 16:46.02 | Drgb_ | cneb3000, thank you very much anyway, I wouldn't have come to the solution without your help |
| 16:46.15 | Drgb_ | offers a virtual beer to cneb3000 |
| 16:55.44 | justdave | so yeah, routing calls over SIP instead of IAX between the two servers seems to eliminate the one-way packet loss problem |
| 16:56.28 | justdave | how strange |
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| 16:57.13 | jpsharp | an ISP mangling/shaping traffic? |
| 17:07.59 | *** join/#asterisk axilla (~axilla@70.89.103.1) |
| 17:08.26 | justdave | The suspicion last night when I was discussing it in here is that the server that was losing its outbound audio was having timing issues. |
| 17:08.47 | axilla | I'm trying to make an extension > extension call in asterisk, but getting a "This number is not in service" message. Asterisk call logs always show "unkown peer > extension" being dialed no matter which direction i call from. |
| 17:08.53 | justdave | it's using dahdi_dummy because there's no physical telephony hardware (the machine itself is a physical box and not a VM though) |
| 17:09.13 | axilla | https://gist.github.com/991375 -- asterisk log |
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| 17:31.03 | justdave | axilla: that looks like you have a default/guest SIP account enabled (which actually doesn't allow any incoming calls, it just plays the invalid message - usually that's a good thing) and your incoming call isn't matching a known SIP login so it's getting accepted by that guest user rather than the one it's supposed to. |
| 17:31.58 | justdave | make sure the line with the [name] in sip.conf matches what the client is using as a username, that's the usual culprit for that kind of mismatch |
| 17:32.39 | *** join/#asterisk DaneoShiga (~danilo.sh@187.39.186.69) |
| 17:33.17 | DaneoShiga | Someone here fluent in DeadAgi? ^^ |
| 17:34.15 | kaldemar | ~ask |
| 17:34.15 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:34.30 | DaneoShiga | I'm executing an agi using deadagi since i want some code to happen after the call is finished, someone knows if there's any reason for it not happen? apparently sometimes the code doesn't execute |
| 17:34.53 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:37.49 | *** join/#asterisk Corazu (2671b70c@gateway/web/freenode/ip.38.113.183.12) |
| 17:42.13 | QuantumSchema | leifmadsen: Sorry, I'm back. |
| 17:42.22 | QuantumSchema | leifmadsen: I got pulled away from my desk. |
| 17:42.42 | QuantumSchema | leifmadsen: So you said I could ignore hints all together for what I'm looking to do? |
| 17:44.06 | irroot | rapture ?? |
| 17:45.38 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 17:46.00 | axilla | justdave: i only have two extensions atm |
| 17:46.04 | axilla | not sure what you mean. |
| 17:46.07 | axilla | 520 and 160 |
| 17:47.16 | justdave | extensions != device name |
| 17:47.21 | justdave | how are the devices named? |
| 17:47.30 | axilla | ohhh |
| 17:47.37 | axilla | I have one named Guesty |
| 17:47.39 | axilla | Guest |
| 17:47.46 | axilla | as it was inteded for a Guest phone |
| 17:48.33 | justdave | it's actually the [general] section of sip.conf coming into play here. |
| 17:48.41 | axilla | i see |
| 17:48.43 | Corazu | Hi there I've got a problem with the FFA and getting Asterisk to read the license. I followed the installation steps and the modules load, except the res_fax_digium throws a warning "Failed to initialize res_fax_digium_copy_protection!" and when I call fax show licenses it doesn't show anything. Does anyone know where I might need to be looking? Google searches haven't yielded me anything so far |
| 17:48.47 | axilla | so i need to change the device name |
| 17:48.51 | justdave | if you have someone connect via SIP and they don't match a known account, they go into the context specified in [general] |
| 17:49.36 | justdave | if you're set up right, that context is a private one that doesn't actually let you do anything (since you don't want random people on the net making unauthenticated long distance calls via your system and so forth) |
| 17:49.53 | justdave | or has a limited set of things available that you're okay with random people from outside hitting |
| 17:50.23 | justdave | you should have a section in sip.conf for each phone device as well |
| 17:50.40 | justdave | whatever you put in [] at the top of each section is the username that device needs to use to connect with |
| 17:51.50 | justdave | a lot of times people will put a context that doesn't actually exist in there, which will cause it to just refuse the call |
| 17:52.00 | justdave | (in the [general] section) |
| 17:52.38 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 17:54.03 | justdave | a context is a named section in your dialplan |
| 17:55.20 | axilla | thanks dave |
| 17:55.25 | axilla | think i figured it out |
| 17:55.29 | axilla | i had something set wrong in my phone config |
| 17:56.31 | *** join/#asterisk nix8n82 (~nate@24.143.28.16) |
| 18:00.09 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:00.09 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:02.58 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust646.15-3.cable.virginmedia.com) |
| 18:06.14 | el3slave | need spell check in data plans :) |
| 18:07.54 | axilla | so it was sending the ipaddress as the ext instead of the ext |
| 18:08.07 | axilla | because polycom labeled the extension in the phone config as Address |
| 18:08.21 | axilla | so i had the ipaddress of the phone there hence the unkown caller... changing that calls started flowing |
| 18:08.23 | axilla | :) |
| 18:10.46 | *** join/#asterisk eerie_ (~mime@newshell1.bshellz.net) |
| 18:17.41 | *** join/#asterisk Daghdha (~a@g239174.upc-g.chello.nl) |
| 18:22.17 | *** join/#asterisk m4xx (~m4xx@75-144-154-165-NewEngland.hfc.comcastbusiness.net) |
| 18:24.41 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 18:25.53 | *** join/#asterisk Cadey (~x@host86-141-160-143.range86-141.btcentralplus.com) |
| 18:29.30 | m_tadeu | is it possible to play an audio file on a channel using ami? |
| 18:31.00 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 18:32.23 | Daghdha | Hi, i have set up asterisk and see i have 2 sip peers. I can't connect to it no matter what i try. the client X-lite says 404-not found. But i don't even see anything on Asterisk console |
| 18:32.28 | Daghdha | Any clues? |
| 18:34.13 | *** join/#asterisk vfabi (~fabi@host-static-188-237-247-62.moldtelecom.md) |
| 18:34.27 | keith4 | crank up the verbosity |
| 18:34.39 | Daghdha | i have it at 11 |
| 18:35.04 | Daghdha | ok that was a cheal spinal tap reference, but it's at 9 |
| 18:35.44 | keith4 | then your client isn't connecting to that asterisk |
| 18:35.58 | Daghdha | is there any information to let the cli say what ports and ip's it's listening on. |
| 18:36.08 | Daghdha | I was affraid of that :/ |
| 18:36.15 | Daghdha | But where is the 404 coming from then. |
| 18:36.22 | atan | Apache? |
| 18:36.52 | Daghdha | I thought it just uses port 5060? |
| 18:37.00 | Daghdha | Apache is on 80, yes. On that machine. |
| 18:37.10 | keith4 | what's your x-lite conf? |
| 18:38.02 | Daghdha | I dunno, i just insatlled it and made an account. |
| 18:39.16 | Daghdha | It doesn't have a config gile |
| 18:41.05 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:41.39 | Daghdha | I see an ;[xlite1] section in sip.conf |
| 18:41.46 | Daghdha | I didn't touch that, it's default |
| 18:43.39 | Cadey | Hi guys, stupid question here but ive never actualy upgraded a version of asterisk. I always install from sauce so am i right in thinking its a simple case of downloading the new tar ball, extract, build and install and that will replace the current version |
| 18:43.45 | Cadey | say fom 1.8 to 18.4 |
| 18:44.01 | *** join/#asterisk h00man (~chatzilla@189.121.241.38) |
| 18:44.12 | h00man | what distro is asterisknow |
| 18:44.22 | leifmadsen | h00man: CentOS |
| 18:44.25 | h00man | thx |
| 18:44.32 | pabelanger | Cadey: yes, remember to restart asterisk too |
| 18:44.44 | Cadey | yeah I would do that part :P |
| 18:44.57 | Daghdha | Ok. asterisk runs, but port 5060 is not used by it. |
| 18:45.12 | h00man | ip-pbx trying out |
| 18:45.56 | Corazu | Hi there I've got a problem with the FFA and getting Asterisk to read the license. I followed the installation steps and the modules load, except the res_fax_digium throws a warning "Failed to initialize res_fax_digium_copy_protection!" and when I call fax show licenses it doesn't show anything. Does anyone know where I might need to be looking? Google searches haven't yielded me anything so far |
| 18:46.36 | leifmadsen | Corazu: that's really an issue you should take up with Digium support directly |
| 18:46.50 | leifmadsen | you're using a commercial module and dealing with commercial things |
| 18:47.01 | Corazu | It's the free version though, they don't support it |
| 18:47.16 | leifmadsen | ah that's true |
| 18:47.35 | Daghdha | If i am trying to use asterisk should a process on my server be listening to port 5060? |
| 18:47.39 | Corazu | everything seems to work except the license key isn't being picked up, I figured someone might have encountered it before |
| 18:48.12 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:51.24 | *** join/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 18:51.25 | otwieracz | Hello. |
| 18:51.48 | otwieracz | I'm trying to setup simple Asterisk SIP server with: http://phplinuxandthelike.wordpress.com/2007/09/04/basic-asterisk-configuration/ |
| 18:51.56 | otwieracz | But, when calling at 55 I'm getting: |
| 18:52.02 | otwieracz | At client: User not found |
| 18:52.05 | otwieracz | At server: [May 25 20:51:13] NOTICE[29488]: chan_sip.c:21581 handle_request_invite: Call from '203' to extension '55' rejected because extension not found in context 'default'. |
| 18:52.57 | otwieracz | (default == home, home had the same problem) |
| 18:53.56 | h00man | is digium still giving proper attention to asterisknow or is their development all going to switchvox |
| 18:56.05 | *** join/#asterisk momobaxter (~dbelrose@web01.derekbelrose.com) |
| 18:56.23 | Daghdha | stop now does nothing here |
| 18:56.32 | Daghdha | Is that something i misconfigured? |
| 18:57.44 | m_tadeu | when I use the ami AGI action, when in time will this action be processed? imagining that the action is sent when the channel is waiting in a queue |
| 18:58.03 | Daghdha | core stop now apparently. tutorial doesn't mention that |
| 18:58.31 | h00man | the whole world is broken. economy, society, politics, culture. can I just reformat and start over? |
| 18:58.47 | carrar | yes |
| 19:00.14 | momobaxter | fdisk, format and reinstall the world |
| 19:02.50 | carrar | You could use this: http://www.tpub.com/neets/book23/0014.GIF |
| 19:03.03 | leifmadsen | nice, just got registration scanned on my asterisk box, and fail2ban didn't catch it because I didn't have a 'Not a local domain' regex filter in filter.d/asterisk.conf |
| 19:03.05 | leifmadsen | I do now :) |
| 19:04.02 | carrar | If you don't get a dozen of those a day you just aren't connected to the internet! :) |
| 19:04.20 | leifmadsen | lol exactly |
| 19:04.26 | Daghdha | are there exploits for asterisk? |
| 19:04.28 | leifmadsen | but fail2ban wasn't banning them -- now they will be :) |
| 19:04.41 | carrar | Asterisk is bullet proof!! |
| 19:04.52 | leifmadsen | asterisk can't be beaten even with a baseball bat! |
| 19:05.00 | sunfone | I have an RBS T1 in an asterisk 1.4.35 / Dahdi 2.3.0 / T4XXP box... lots of remote SIP peers (Linksys PAP2T/2102)... Often I see Dahdi channels in state "Ringing..." forever, obviously abandoned by the SIP peer. Eventually they go away, but through the day I accumulate many. A "soft hangup" on the channel kills it, and I have been considering a script to recognize such a channel and hang it up this way, but would like to better unders |
| 19:05.01 | leifmadsen | asterisk can't be killed with conventional weapons! |
| 19:05.18 | carrar | leifmadsen, put it on your blog! |
| 19:05.40 | Daghdha | It just sits like a vegetable on my console tbh. |
| 19:05.44 | carrar | the f2b stuff |
| 19:05.57 | leifmadsen | carrar: I should probably update a wiki somewhere too... |
| 19:06.03 | carrar | yeah |
| 19:06.06 | carrar | wiki that pls k thanks |
| 19:06.52 | carrar | http://en.wikipedia.org/wiki/leifmadsen_documents_fail2ban |
| 19:06.59 | leifmadsen | :) |
| 19:07.09 | leifmadsen | brb |
| 19:07.24 | Daghdha | when a phone tries to register what port doe sit use? |
| 19:07.35 | carrar | SIP? |
| 19:07.39 | sunfone | by default 5060 |
| 19:07.44 | sunfone | UDP |
| 19:08.11 | Daghdha | u have no process listening on that port. Asterisk is running and X-LIte says 404. |
| 19:08.43 | Daghdha | where does it get the 404 from? Asterisk isn't listening on 5060 |
| 19:08.51 | otwieracz | Which module I need to parse extensions.conf? |
| 19:09.17 | h00man | any know a faq/tutorial/howto for setting up asterisknow into a standard smalloffice PBX |
| 19:09.20 | sunfone | 404 is the return code for a failed registration |
| 19:09.36 | sunfone | asterisk must have the SIP module loaded |
| 19:09.43 | sunfone | to listen on 5060 |
| 19:10.02 | Daghdha | who gives my phone the 404 though? |
| 19:10.03 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 19:10.33 | sunfone | Why do you think asterisk is not listening on 5060? |
| 19:10.52 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 19:10.53 | h00man | sip module not installed default? |
| 19:10.56 | Daghdha | coz netstat and lsof -i don't show it |
| 19:12.40 | sunfone | what is the output of "lsof -i | grep sip" |
| 19:12.49 | Daghdha | when i start asterisk it stops with "Asterisk ready." and a blinking cursor below the A |
| 19:13.24 | Daghdha | sipw 1545 root 8u IPv4 9388 0t0 UDP *:sip |
| 19:13.52 | sunfone | so you are running asterisk as the user "sipw", and it is indeed listening on 5060 (sip) |
| 19:14.03 | Daghdha | DOH |
| 19:14.08 | sunfone | :) |
| 19:14.58 | h00man | whats the best distro for asterisk/linux newbies |
| 19:14.59 | QuantumSchema | What exactly does the "stateinterface" option in "AddQueueMember" do for me? (it's the last option in the command) |
| 19:17.13 | h00man | hmm ok asterisknow boots and lets me reach freepbx admin, what now |
| 19:18.20 | QuantumSchema | better yet.. .does anyone know if this patch (https://issues.asterisk.org/view.php?id=15168) has made it into 1.8 by now? |
| 19:19.07 | Daghdha | sunfone: So you are saying asterisk sends the 404? But why don't i see any output on my vvvvv cli? |
| 19:20.14 | h00man | is a "trunk" the same as a SIP-voip-provider consumer "phone line"? |
| 19:21.19 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:21.19 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:21.50 | h00man | hmm ok so a trunk is supposed to have many "lines" within |
| 19:22.46 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 19:24.11 | sunfone | A "SIP trunk" is more analogous to a TDM T1 than a POTs line, if that is what you are asking |
| 19:24.47 | Daghdha | or: yes |
| 19:25.35 | sunfone | Some people will get upset if you use the word "trunk" and "sip" together though :) |
| 19:25.47 | WIMPy | Or a "sip trunk" is a term that doesn't really make sense. Just that ppl are used to call theyr default routes trunks. |
| 19:26.00 | sunfone | heh... good timing :) |
| 19:26.50 | Daghdha | My mom says my dad's a trunk |
| 19:26.56 | Daghdha | He likes a sip now and then |
| 19:27.15 | sunfone | FreePBX has muddied those waters - calling all such connections "trunks" |
| 19:27.27 | Daghdha | On that bombshell.. i deleted all that is asterisk an wish you a pleasent continuation. Bye |
| 19:27.30 | *** part/#asterisk Daghdha (~a@g239174.upc-g.chello.nl) |
| 19:32.19 | h00man | well it seems the SIP providers only only for one simultaneous call over a particular account usually, thereby making it similar to a "single line" |
| 19:33.07 | h00man | got a $10 inphonex account to try out yesterday, see they have a different account for asterisk-trunks |
| 19:33.33 | sunfone | its all about how many simultaneous external conversations you want |
| 19:33.50 | sunfone | for $10/month you probably only get one |
| 19:34.07 | sunfone | (inbound) |
| 19:34.14 | momobaxter | are there any ami experts around that want to tell me I'm crazy for trying something? |
| 19:34.25 | sunfone | you are crazy for trying something |
| 19:34.30 | momobaxter | sweet. |
| 19:34.31 | momobaxter | thanks :) |
| 19:34.33 | sunfone | :) |
| 19:34.57 | WIMPy | You are trying something? You must be crazy! |
| 19:35.14 | sunfone | Ok, I'll bite. What are you trying? |
| 19:36.05 | h00man | uhm $10 prepaid voip dialout from inphonex... |
| 19:37.12 | sunfone | odd for them to limit outbound |
| 19:37.21 | momobaxter | ok, i have at any point 40 outgoing reps here. At some point in the day, someone will need to basically do the following: Park the call, Call a third party and bring the 3 together into a Meetme. I need this as idiot proof as possible, meaning in the middle of the call hit *444 or something, it prompt for the number and hit # again and brings them all together into a conference channel |
| 19:37.38 | sunfone | most termination providers, IMO, will let you call as much as you want - you just use up your balance faster :) |
| 19:38.10 | sunfone | momo: yup that sounds like an AMI app |
| 19:38.41 | sunfone | momo: I did something similar with AGI a year ago... and call files... but AMI would probably be cleaner |
| 19:38.42 | momobaxter | so I subscribe to DTMF events, wait for a particular series of them on a channel and go from there? |
| 19:39.20 | sunfone | I don't know that you need to monitor DTMF events |
| 19:39.25 | WIMPy | I think it could be done with dynamic features as well. |
| 19:39.45 | sunfone | features.conf would be the way to trap the 444 |
| 19:39.55 | sunfone | have that enter the dialplan |
| 19:40.12 | momobaxter | i tried that alrady with a simple ael script |
| 19:40.16 | sunfone | then your AGI or whatever could handle the meetme creation |
| 19:40.30 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:40.44 | momobaxter | the problem I ran into is that dynamic features only work on the caller or the callee...this needs both |
| 19:40.59 | momobaxter | unless I'm thinking of this incorrectly. |
| 19:41.35 | sunfone | [re-reading you explanation...] |
| 19:41.43 | momobaxter | peer vs self |
| 19:41.48 | momobaxter | sorry, wrong terminology |
| 19:42.10 | sunfone | isn't it your agent that needs to do the parking and conferencing? |
| 19:42.19 | sunfone | you want the caller to be able to do it also? |
| 19:42.22 | momobaxter | no. |
| 19:42.29 | momobaxter | agent = caller |
| 19:42.34 | momobaxter | customer = callee |
| 19:42.36 | momobaxter | we're outbound |
| 19:42.39 | sunfone | ahh |
| 19:43.10 | sunfone | I don't see why features won't work for the caller... I think it is enabled by a DIAL option |
| 19:43.24 | sunfone | (off the top of my head, admittedly) |
| 19:43.31 | momobaxter | Like I said I wrote a simple ael script that just ran park |
| 19:43.46 | momobaxter | so when the caller hit #444 it ended up parking the caller and disconnecting the callee |
| 19:44.00 | sunfone | heh |
| 19:44.04 | sunfone | I bet you were popular then |
| 19:44.16 | h00man | asterisknow flash-operator-panel doesnt seem to work |
| 19:44.55 | sunfone | Do you really need to park it? Don't you just want to put them on hold? |
| 19:46.58 | sunfone | At any rate you had the right approach already - no need to make it more complicated looking for DTMF and reproducing the dynamic features functionality. |
| 19:47.28 | sunfone | You would end up have the same trouble when trying to park "the new way", and should focus on figuring out why it dropped the other channel |
| 19:47.52 | sunfone | if parking is really what you want to do... but it seems that you really just want to create another channel and bridge them |
| 19:48.05 | sunfone | so just playing some hold music for the original channel may be the right route |
| 19:48.57 | *** join/#asterisk momobaxter (~dbelrose@web01.derekbelrose.com) |
| 19:48.59 | momobaxter | ugh |
| 19:49.06 | momobaxter | hurray for timeouts |
| 19:49.41 | otwieracz | [May 25 21:49:31] WARNING[30746]: file.c:644 ast_openstream_full: File demo-echotest does not exist in any format |
| 19:50.09 | otwieracz | [root@linvoip asterisk]# stat /var/lib/asterisk/sounds/en/demo-echotest.gsm File: `/var/lib/asterisk/sounds/en/demo-echotest.gsm' |
| 19:50.20 | otwieracz | [May 25 21:49:31] WARNING[30746]: file.c:950 ast_streamfile: Unable to open demo-echotest (format 0x2 (gsm)): Resource temporarily unavailable |
| 19:50.28 | otwieracz | Why? |
| 19:51.40 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 20:01.47 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:03.04 | GreatSUN | re |
| 20:05.52 | momobaxter | .quit |
| 20:06.50 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 20:07.22 | h00man | yum update |
| 20:15.20 | jaytee | ~botsnack |
| 20:15.20 | infobot | aw, gee, jaytee |
| 20:15.39 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 20:16.30 | *** join/#asterisk pdtpatrick_ (~pdtpatric@mainstwan.farheap.com) |
| 20:17.26 | sezuan | If I use ldap for sip.conf, will the entries in sip.conf ignored or merged with the ldap data? |
| 20:21.39 | *** join/#asterisk phoenixsampras (~phoenixsa@static-190-181-38-119.acelerate.net) |
| 20:21.54 | phoenixsampras | help, how to know if the FXO card is working? |
| 20:32.07 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 20:32.09 | pa | hi |
| 20:32.10 | pa | sorry |
| 20:32.14 | pa | i have a stupid question |
| 20:32.50 | jaytee | ~ask |
| 20:32.51 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:33.00 | pa | how should i create an extension to call an user listed in the sip.conf? |
| 20:33.41 | jaytee | exten => somenumber,1,Dial(SIP/YOURACCOUNT) |
| 20:34.23 | pa | cant i have something with wildcard? |
| 20:34.35 | pa | like username,1,Dial(SIP/username) |
| 20:34.36 | jaytee | sure |
| 20:35.27 | pa | do i need some special character? |
| 20:35.31 | pa | dollar or somethin? |
| 20:35.34 | jaytee | I have several extensions that all begin with 2 so I have a pattern match exten => _2XX,1,Dial(SIP/${EXTEN}) but my sip accounts match their extensions |
| 20:35.57 | pa | ah i understand |
| 20:36.21 | pa | so in my case i need something like 7_,1,Dial(SIP/${EXTEN}) |
| 20:36.22 | jaytee | the ${EXTEN} is the asterisk variable for the dialed extension. |
| 20:36.30 | jaytee | no |
| 20:36.35 | pa | and then i call 7user, and i get SIP/usr? |
| 20:36.38 | pa | user |
| 20:36.42 | jaytee | the _ indicates a pattern match and has to be first |
| 20:37.12 | pa | hmm |
| 20:40.59 | jaytee | since most telephones cannot dial letters you can't directly dial an alphabetic SIP account. You'd need to map it in your dialplan to a dialable number |
| 20:41.09 | pa | ah thanks, it worked :) |
| 20:41.18 | jaytee | ~book |
| 20:41.18 | infobot | For more information about the Asterisk book, see ~thebook |
| 20:41.25 | jaytee | ~thebook |
| 20:41.25 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 20:41.53 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 20:42.17 | jaytee | pa, if you're just starting I'd highly recommend you check that out. It's chock full o' goodies and is the 3rd edition of what most people using Asterisk consider to be "the Bible" |
| 20:43.08 | pa | i+ll check it out, it seems free :) |
| 20:43.20 | jaytee | free to read online |
| 20:44.48 | jaytee | but if you and 300 others buy it the authors might actually be able to buy that X-Box 360 they've been drooling over. :-) |
| 20:47.10 | h00man | outbound route or trunk for a SIP dialout provider? |
| 20:47.16 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 20:47.33 | jaytee | is that a question? |
| 20:47.57 | jaytee | are you looking for a list of SIP providers? |
| 20:48.49 | h00man | signed up for a sip provider, a dial-out account - inphonex. trying to figure out where to add it in asterisknow 1.7.1 |
| 20:49.39 | jaytee | it would be under trunks I believe. been awhile since I used *Now |
| 20:49.46 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:50.04 | jaytee | but it will give you a choice of SIP, IAX2 and something else IIRC. |
| 20:50.53 | h00man | I have SIP login details for it. |
| 20:51.03 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:54.06 | jaytee | h00man, did you choose the Asterisk GUI or the FreePBX option? |
| 20:54.11 | h00man | "trunks" > "add SIP trunk" gives a dialog asking for "PEER details" and "USER details" |
| 20:54.24 | h00man | I'm in the FreePBX interface |
| 20:54.52 | jaytee | try asking in #freepbx channel. most people in here use standard vanilla asterisk without a gui |
| 20:55.20 | h00man | i see |
| 20:55.22 | jaytee | so we're not as familiar with setting things up in a web interface. the gui of choice for most in this channel is vi |
| 20:55.28 | jaytee | or in my case nano |
| 20:55.35 | h00man | yes makes sense |
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| 20:56.03 | ChannelZ | vomits on vi |
| 20:56.18 | h00man | sorry for invading with gui questions then |
| 20:56.55 | jaytee | no problem, just thought you'd get a clearer answer and advice there than in here. I'd help if I knew the gui better. |
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| 20:59.26 | pdtpatrick_ | question .. if i have a configuration on another box .. how can i have my current box to pick up a call and send it to the other box.. im trying SIP/82782 but it does not see that since it is not local |
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| 21:00.32 | h00man | I you edit the proper conf file does freepbx usually read the file properly? |
| 21:00.50 | phoenixsampras | Help!! Asterisk is not hunging up outgoing calls |
| 21:00.58 | pa | i have another question: does asterisk support video over SIP? |
| 21:01.00 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 21:01.07 | pa | (out of the box) |
| 21:01.14 | jaytee | pa, not straight out of the box |
| 21:01.36 | jaytee | you need to add some statements in sip.conf for asterisk, not sure where in AsteriskNOW |
| 21:01.42 | pa | ah |
| 21:01.46 | pa | like what statement? |
| 21:02.16 | jaytee | videosupport=yes in the general section of sip.conf and allow=h323 in the codecs section |
| 21:02.48 | jaytee | oops, got that allow wrong |
| 21:03.53 | jaytee | allow=h263 |
| 21:03.57 | h00man | though asterisknow was the same as plain asterisk, with freepbx and centos? |
| 21:04.05 | h00man | I thouhgt |
| 21:04.21 | h00man | or perhaps I thought. |
| 21:05.00 | jaytee | kind of, but using a database and extra config files puts constraints on the system and means you can't always directly edit a configuration file. |
| 21:05.34 | jaytee | for most purposes that isn't a problem but if you need a customized system with specialized functions the gui gets in the way |
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| 21:08.53 | pa | last question :) is there some app for iphone that allows SIP video calls? |
| 21:09.06 | GreatSUN | h00man: asterisknow != asterisk |
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| 21:09.35 | pa | i use xlite, and here it seems video call are supported |
| 21:09.52 | pa | but on windows |
| 21:09.55 | pa | not iphone |
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| 21:10.02 | GreatSUN | h00man: asterisknow is a distribution for newbies while asterisk itself is just the plain software you have to configure by editing config-files and constructing dial plans |
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| 21:12.22 | Freeaqingme | GreatSUN: if asterisknow is for newbies, what would freepbx be? |
| 21:12.47 | jaytee | quittin time, I'm out. peace all! |
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| 21:13.55 | GreatSUN | Freeaqingme: alike |
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| 21:26.39 | m_tadeu | I'm having a problem with async agi....my AMI action for an agi command fails with this message "Channel SIP/213.63.185.58:1720-00000002 does not exists or cannot get its lock" |
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| 21:31.40 | h00man | 3cx zip softphone on windows also has video support. |
| 21:31.56 | h00man | mean 3cx sip softphone |
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| 22:46.52 | MrNemus | so what would stop sip channels from closing ? |
| 22:48.17 | MrNemus | http://pastebin.com/UhkJeBkf this is what i see and the sip channels keep growing |
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| 23:03.58 | emsLinux | Hello, I got this problem... I'm using Asterisk 1.8.4 over Ubuntu Server with FreePBX 2.9.. Everything seems to run OK, but it is not. I'm trying to connect several softphones and hardphones and i cant make it work. If they got registered when I try to make a call, nobody listens anybody, but i don't understand what is going on, everything seems to be ok. |
| 23:03.59 | emsLinux | This problem rings any bell on anybody who can help me? Anyone knows what could be possibly wrong? |
| 23:06.02 | paulc | emsLinux: I think you need #freepbx - the crowd in here are more versed in "traditional" Asterisk configuration |
| 23:07.39 | emsLinux | but i first tried to make it work only with asterisk and i couldn't, the same problem, also the server is only for sip conections, not LAN, just Internet |
| 23:09.49 | ChannelZ | I don't even understand the problem. Do calls work, you just get no audio? |
| 23:11.24 | emsLinux | no audio and got a lot of problems conecting devices, with several IPs and systems |
| 23:11.30 | citywok | if they are for internet... are you sure you have the right ports forwarded to asterisk? |
| 23:11.43 | emsLinux | iPhones, softphones for PC, even one ATA and a IP Phone |
| 23:11.52 | citywok | the RTP media ports are important, if they aren't allowed then you won't get any audio |
| 23:12.01 | emsLinux | 0-65353 udp and tcp, nothing |
| 23:12.45 | ChannelZ | are these phones behind a firewall/NAT? |
| 23:13.23 | emsLinux | i'am so frustrated, i think could be the latest version of Asterisk, maybe a modulo, but im not that good using asterisk to understand what is happening |
| 23:13.40 | citywok | emsLinux: does it work with 2 phones on the same lan as *? |
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| 23:14.03 | emsLinux | they are in several places, some of them behind NAT, but they worked before with another version of Asterisk with no problem |
| 23:14.18 | citywok | do 2 phones without nat work right now? |
| 23:14.49 | emsLinux | Cant, is a cloud server, but we have another cloud server working fine, this doesnt want to =( |
| 23:15.49 | citywok | make a packetcapture on the external interface and make a test call, then open the packet capture in wireshark and see if you see both legs of the call |
| 23:16.19 | emsLinux | ok |
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| 23:34.40 | mikeyutley | question so do time conditions apply to the whole system? |
| 23:35.03 | mikeyutley | or can you apply them specific extensions? |
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| 23:38.43 | mikeyutley | anybody? |
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| 23:40.18 | nny | looking at docs for 3.3.1 polycoms, anyone know if/how presence is enabled (besides directory.xml)? There used to be an enable/dsiable switch in the config files universally and I can't find the entry in the admin guide |
| 23:41.27 | nny | nm sry found a good forum post |
| 23:41.36 | nny | ha well nm |
| 23:41.47 | nny | "Followup: After not being able to figure out what was wrong, I tried downgrading to 3.2.5. All is well now... so I'm leaving it alone until I have more time to mess with it." <---- crap! |
| 23:45.29 | nny | well.. actually it's here in the admin guide. time to test |
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